In many people who are profoundly deaf, the reason for deafness is absence of, or destruction of, the hair cells in the cochlea which transduce acoustic signals into nerve impulses. These people are thus unable to derive suitable benefit from conventional hearing aid systems, no matter how loud the acoustic stimulus is made, because there is damage to or absence of the mechanism for nerve impulses to be generated from sound in the normal manner.
It is for this purpose that cochlear implant systems have been developed. Such systems bypass the hair cells in the cochlea and directly deliver electrical stimulation to the auditory nerve fibres, thereby allowing the brain to perceive a hearing sensation resembling the natural hearing sensation normally delivered to the auditory nerve. U.S. Pat. No. 4,532,930, the contents of which are incorporated herein by reference, provides a description of one type of traditional cochlear implant system.
Typically, cochlear implant systems have consisted of essentially two components, an external component commonly referred to as a processor unit and an internal implanted component commonly referred to as a stimulator/receiver unit. Traditionally, both of these components have cooperated together to provide the sound sensation to a user.
The external component has traditionally consisted of a microphone for detecting sounds, such as speech and environmental sounds, a speech processor that converts the detected sounds, particularly speech, into a coded signal, a power source such as a battery, and an external transmitter coil.
The coded signal output by the speech processor is transmitted transcutaneously to the implanted stimulator/receiver unit situated within a recess of the temporal bone of the user. This transcutaneous transmission occurs via the external transmitter coil which is positioned to communicate with an implanted receiver coil provided with the stimulator/receiver unit. This communication serves two essential purposes, firstly to transcutaneously transmit the coded sound signal and secondly to provide power to the implanted stimulator/receiver unit. Conventionally, this link has been in the form of an RF link, but other such links have been proposed and implemented with varying degrees of success.
The implanted stimulator/receiver unit traditionally includes a receiver coil that receives the coded signal and power from the external processor component, and a stimulator that processes the coded signal and outputs a stimulation signal to an intracochlea electrode assembly which applies the electrical stimulation directly to the auditory nerve producing a hearing sensation corresponding to the original detected sound.
Traditionally, the external componentry has been carried on the body of the user, such as in a pocket of the user's clothing, a belt pouch or in a harness, while the microphone has been mounted on a clip mounted behind the ear or on the lapel of the user.
More recently, due in the main to improvements in technology, the physical dimensions of the speech processor have been able to be reduced allowing for the external componentry to be housed in a small unit capable of being worn behind the ear of the user. This unit allows the microphone, power unit and the speech processor to be housed in a single unit capable of being discretely worn behind the ear, with the external transmitter coil still positioned on the side of the user's head to allow for the transmission of the coded sound signal from the speech processor and power to the implanted stimulator unit.
In earlier versions of speech processors, the processor used feature extraction strategies to identify the speech features present in the signal from the microphone and encode them as patterns of electrical stimulation. Typically, the features of the speech that were extracted were the fundamental frequency (or voice pitch) and the amplitudes and frequencies of the first and second formants of the speech spectrum. Such processing had the advantage that the hardware required to perform the feature extraction could be relatively simple so leading to a relatively low power consumption. Strategies that employed this feature extraction philosophy were found to work particularly well when the user was listening to a single voice in a quiet environment, however, when the user was in an environment with background noise the strategy was not nearly as successful. If, for example, two people were speaking at the same time, then two first formants would be mixed. The processor in expecting only one formant provided a single estimate of this formant which was a mixture of the two. The result was a signal which the user could not readily understand.
A new approach was subsequently developed that provided a full range of spectral information without any attempt by the hardware to fit it into a preconceived mould. The user was then given an opportunity to listen for the particular information of interest and identify the speech features themselves, in the presence of the background noise. In this approach, the overall sound spectrum is analysed and divided into a number of frequency bands with the electrodes stimulated in a tonotopic fashion according to the energy in those bands. This has a number of advantages as it saves power, allows a higher stimulation rate to be employed since time is not wasted in presenting unimportant stimuli, and also serves to decrease the annoyance of background noise.
Although there are differences between speech processors for different cochlear implants and also speech processors used in hearing aid applications, there are also many common features. A speech processor firstly typically includes a preamplifier and automatic gain control (AGC). The preamplifier amplifies the very low signal detected from the microphone to a suitable level that can be handled by the rest of the speech processor. The AGC controls the level of the signal so that it does not overload or distort. The AGC can have what is known as infinite compression in that the signal is amplified by a fixed gain until the output signal reaches a certain maximum level, at which the gain is reduced to prevent the output signal from exceeding that level. For example, the gain may be controlled in order to ensure that an output signal never exceeds a maximum comfort value for the user.
It has been found that users of cochlear implant systems that have an automatic gain control (AGC) tend to set the sensitivity to a level such that the AGC does not enter infinite compression except at high input signal levels, such as when they themselves speak. The motive for this is that setting the sensitivity higher means that the gain is reduced during speech that the user wants to hear, but is increased when the speaker stops, thus amplifying the background noise. Setting the sensitivity lower results in some of the signal falling outside the stimulation range, and so reducing speech perception. In summary, patients set the sensitivity control to maximise the perceived signal to noise ratio, ie. the ratio between speech and background noise in the absence of speech. In general, the sensitivity control is set so that the background noise is not too obtrusive.
A problem can occur with this system when a user is faced with an environment where the level of background noise is varying. To address this problem, an Automatic Sensitivity Control (ASC) has been devised. The ASC controls the background noise level by constantly monitoring the signal from the microphone and recording the minimum level to which it drops over a period of several seconds (generally 5-10 seconds). This minimum level is called the noise floor. The ASC adjusts the gain so that the noise floor is held below a predetermined breakpoint, usually so that the user's threshold hearing level corresponds approximately to the noise floor. The gain sensitivity adjustment may be made manually or by an automatic means such as is described in International Publication No WO 96/13096, the contents of which are incorporated by reference. Although this system provides improved listening comfort for the user, the system does have the disadvantage that at low speech levels, a step of simply linearly increasing the amount of gain is insufficient to maintain such speech perception at a satisfactory level.
The present inventors have recognised the shortcomings of current hearing device sensitivity control techniques and practices in the prior art and accordingly have sought to provide an improved system and method of controlling the sensitivity of hearing devices, such as cochlear implants.
Any discussion of documents, acts, materials, devices, articles or the like which has been included in the present specification is solely for the purpose of providing a context for the present invention. It is not to be taken as an admission that any or all of these matters form part of the prior art base or were common general knowledge in the field relevant to the present invention as it existed before the priority date of each claim of this application.
Throughout this specification the word “comprise”, or variations such as “comprises” or “comprising”, will be understood to imply the inclusion of a stated element, integer or step, or group of elements, integers or steps, but not the exclusion of any other element, integer or step, or group of elements, integers or steps.